Modern telecommunications continue to evolve toward ever increasing use of the Internet for data transmissions. In support of this evolution, myriad traffic format protocols have been defined and implemented to enable transmission of various types of data for defined purposes. Among these protocols is the session initiation protocol (SIP), which is generally used in order to route and initiate the transmission of telecommunications of various types of data. Although SIP is capable of initiating the transmission of data of any type, SIP is particularly useful and desirable for initiating transmission of live communications. In particular, SIP is a preferred protocol for initiating the transmission of live two way voice communications such as telephone calls. As the drive for reduced costs and higher signal quality relentlessly grows, maximizing the quality of SIP initiation of transmission of live voice communications over the Internet is particularly desirable.
Live voice communications in particular demand minimal delays in and failures of initiation of call completion. The quality standards set long ago for time division multiplexing protocols and other systems for transmitting live voice communications must likewise be met for Internet based routing to be competitive. Internet communications depend on remotely located servers to route and initiate the transmission of a given signal through the web of potential pathways constituting the Internet. If a server fails during initiation of transmission of a live voice communication, the conversation may be delayed or never connected. Desirably, provisions are accordingly made to minimize service disruptions resulting from a termination of support for initiation of such communications.
SIP protocol communications overlay the concurrent implementation of stream control transmission protocol (SCTP). SCTP is one of many general purpose communications protocols. SIP protocol communications can provide for initiation of transmissions of voice over Internet protocol (VoIP). Although SIP is particularly useful for initiating transmission of live two way telecommunications such as telephone conversations, SIP does support and can be used for initiating transmission of other live or off line data communications.
SCTP includes a three way handshake shutdown protocol for shutting down SCTP server associations with clients. However, the SCTP handshake shutdown protocol does not provide a means for the server to stop or reject new SIP service invites received from clients while the server completes processing of requests that the server has already received and accepted. As a result, shutting down the SCTP associations prevents the server from responding to some service requests already received, possibly causing a loss of or delay in the calls associated with those requests. Moreover, support for SIP protocol communications cannot be shut down separately from shutdown of all SCTP protocol support.
There is accordingly a need for discrete systems and methods to manage terminations of support for SIP protocol communications, in order to minimize service interruptions and provide improved telecommunications quality.